In the field of telephony communication, there have been many improvements in technology over the years that have contributed to more efficient use of telephone communication within hosted call-center environments. Most of these improvements involve integrating the telephones and switching systems in such call centers with computer hardware and software adapted for, among other things, better routing of telephone calls, faster delivery of telephone calls and associated information, and improved service with regards to client satisfaction. Such computer-enhanced telephony is known in the art as computer-telephony integration (CTI).
Generally speaking, CTI implementations of various design and purpose are accomplished both within individual call-centers and, in some cases, at the network level. For example, processors running CTI software applications may be linked to telephone switches, service control points (SCP), and network entry points within a public or private telephone network. At the call-center level, CTI-enhanced processors, data servers, transaction servers, and the like, are linked to telephone switches and, in some cases, to similar CTI hardware at the network level, often by a dedicated digital link. CTI and other hardware within a call-center are commonly referred to as customer premises equipment (CPE). It is the CTI processor and application software at such centers that provides computer enhancement to a call center.
In a CTI-enhanced call center, telephones at agent stations are connected to a central telephony switching apparatus, such as an automatic call distributor (ACD) switch or a private branch exchange (PBX). The agent stations may also be equipped with computer terminals such as personal computer/video display unit's (PC/VDU's) so that agents manning such stations may have access to stored data as well as being linked to incoming callers by telephone equipment. Such stations may be interconnected through the PC/VDUs by a local area network (LAN). One or more data or transaction servers may also be connected to the LAN that interconnects agent stations. The LAN is, in turn, connected to the CTI processor, which is connected to the call switching apparatus of the call center.
When a call arrives at a call center, whether or not the call has been pre-processed at an SCP, typically at least the telephone number of the calling line is made available to the receiving switch at the call center by the network provider. This service is available by most networks as caller-ID information in one of several formats such as Automatic Number Identification Service (ANIS). If the call center is computer-enhanced (CTI) the phone number of the calling party may be used to access additional information from a customer information system (CIS) database at a server on the network that connects the agent workstations. In this manner information pertinent to a call may be provided to an agent, often as a screen pop.
In recent years, advances in computer technology, telephony equipment, and infrastructure have provided many opportunities for improving telephone service in publicly-switched and private telephone intelligent networks. Similarly, development of a separate information and data network known as the Internet, together with advances in computer hardware and software have led to a new multi-media telephone system known in the art by several names. In this new systemology, telephone calls are simulated by multi-media computer equipment, and data, such as audio data, is transmitted over data networks as data packets. In this application the broad term used to describe such computer-simulated telephony is Data Network Telephony (DNT).
For purposes of nomenclature and definition, the inventors wish to distinguish clearly between what might be called conventional telephony, which is the telephone service enjoyed by nearly all citizens through local telephone companies and several long-distance telephone network providers, and what has been described herein as computer-simulated telephony or DNT. The conventional system is familiar to nearly all, and is often referred to in the art as connection-oriented-switched-telephony (COST). The COST designation will be used extensively herein. The computer-simulated, or DNT systems are familiar to those who use and understand computer systems. Perhaps the best example of DNT is telephone service provided over the Internet, which will be referred to herein as Internet-Protocol-Network-Telephony (IPNT), by far the most extensive, but still a subset of DNT.
Both systems use signals transmitted over network links. In fact, connection to data networks for DNT such as IPNT is typically accomplished over local telephone lines, used to reach such as an Internet Service Provider (ISP). The definitive difference is that COST telephony may be considered to be connection-oriented telephony. In the COST system, calls are placed and connected by a specific dedicated path, and the connection path is maintained over the time of the call. Bandwidth is thus assured. Other calls and data do not share a connected channel path in a COST system. In a DNT system, on the other hand, the system is not dedicated or connection oriented. That is, data, including audio data, is prepared, sent, and received as data packets. The data packets share network links, and may travel by variable paths, being reassembled into serial order after receipt. Therefore, bandwidth is not guaranteed.
Under ideal operating circumstances a DNT network, such as the Internet, has all of the audio quality of conventional public and private intelligent telephone-networks, and many advantages accruing from the aspect of direct computer-to-computer linking. However, DNT applications must share the bandwidth available on the network in which they are traveling. As a result, real-time voice communication may at times suffer dropout and delay. This is at least partially due to packet loss experienced during periods of less-than-needed bandwidth which may prevail under certain conditions such as congestion during peak periods of use, and so on.
Recent improvements to available technologies associated with the transmission and reception of data packets during real-time DNT communication have enabled companies to successfully add DNT, principally IPNT capabilities, to existing CTI-enhanced call centers. Such improvements, as described herein and known to the inventor, include methods for guaranteeing available bandwidth or quality of service (QoS) for a transaction, improved mechanisms for organizing, coding, compressing, and carrying data more efficiently using less bandwidth, and methods and apparatus for intelligently replacing lost data by using voice supplementation methods and enhanced buffering capabilities.
In typical call centers, DNT is accomplished by Internet connection and IPNT calls. For this reason, IPNT and the Internet will be used almost exclusively in examples to follow. It should be understood, however, that this usage is exemplary, and not limiting.
In systems known to the inventors, incoming IPNT calls are processed and routed within an IPNT-capable call center in much the same way as COST calls are routed in a CTI-enhanced center, using similar or identical routing rules, waiting queues, and so on, aside from the fact that there are two separate networks involved. Call centers having both CTI and IPNT capability utilize LAN-connected agent-stations with each station having a telephony-switch-connected headset or phone, and a PC connected, in most cases via LAN, to the LAN over which IPNT calls may be routed. Therefore, in most cases, IPNT calls are routed to the agent's PC while conventional telephony calls are routed to the agent's conventional telephone or headset. However, a method known to the inventor allows one headset to be used at an agent's station for handling both IPNT and COST calls. This is accomplished via connecting the agent's telephone to the sound card on the agent's PC/VDU with an I/O cable. In most prior art and current art systems, separate lines and equipment must be implemented for each type of call weather COST or IPNT.
Due in part to added costs associated with additional equipment, lines, and data ports that are needed to add IPNT capability to a CTI-enhanced call-center, companies are currently experimenting with various forms of integration between the older COST system and the newer IPNT system. For example, by enhancing data servers, interactive voice response units (IVRs), agent-connecting networks, and so on, with the capability of understanding Internet protocol, data arriving from either network may be integrated requiring less equipment and lines to facilitate processing, storage, and transfer of data. However, telephony trunks and IPNT network lines representing the separate networks involved still provide for significant costs and maintenance.
In some current art implementations, incoming data from the COST network and the Internet is caused to run side by side from the network level to a call center over a telephone connection (T1/E1) acting as a telephone-data bridge, wherein a certain channels are reserved for COST connection, and this portion is dedicated as is necessary in COST protocol (connection oriented), and the remainder is used for DNT such as IPNT calls, and for perhaps other data transmission. Such a service is generally offered by a local phone company. This service eliminates the requirement for leasing numerous telephony trunks and data-network connections. Routing and other equipment, however, must be implemented at both the call-center level and network level significantly reducing any realized cost savings.
A significant disadvantage of such a bridge, having dedicated equipment on each end, is the dedicated nature of individual channels over the bridging link. Efficient use of bandwidth cannot be assured during variable traffic conditions that may prevail at certain times. For example, dedicated channels assigned to IPNT traffic would not be utilized if there were not enough traffic to facilitate their use. Similarly, if there was more COST traffic than the allotted number of COST channels could carry, no additional channels could be made available.
In a yet more advanced system, known in some call centers, a central switch within the call center is enhanced with IP conversion capability and can communicate via LAN to connected IP phone-sets and PC's eliminating the need for regular telephone wiring within a call center. However, the service is still delivered via a telephone-data bridge as described above. Therefore, additional requirements for equipment and inefficiency regarding use of bandwidth are still factors.
In still other systems known to the inventor, IPNT to COST conversion or COST to IPNT conversion is performed within the call center instead of via a network bridge. This is accomplished via a gateway connected to both an IPNT router and a central telephony-switching apparatus. In the first case, all calls are converted to and routed as COST calls over internal telephone wiring to switch-connected headsets. In the second case, all COST calls are converted to and routed as IPNT calls over a LAN to individual PC/VDU's.
In all of the described prior art systems, the concerted goal has been to integrate COST and IPNT data via converging at the network level or within the call center. The addition of dedicated hardware both at the network level and within the call center adds to the expense of providing such integrated data.
In a system known to the inventor and described with reference to Ser. No. 09/160,558 listed in the cross-reference section of this specification, an integrated router is provided within a call center. The integrated router monitors and controls both a telephony switch receiving and forwarding connection-oriented, switched telephony (COST) calls and a Data Network Telephony (DNT) processor receiving and forwarding DNT calls. The integrated router is enabled by software to consult a common data repository storing status of agents answering both types of calls within the center and routes all calls according to a single set of routing rules, which can take a variety of forms. In one aspect, telephone devices at agent stations are adapted to handle both COST and DNT calls.
It has occurred to the inventor that in addition to being able to unify all routed events within a communication center under a common set of rules, it would be desirable to adapt established IP network protocols for use as routing tools within a communication center for the purpose of saving time and costs of developing proprietary protocols and expensive client applications using them.
One standard Internet-based protocol that may be adapted for communication center use is the well-known session initiation protocol (SIP). Very basically, SIP is an application-layer control (signaling) protocol for creating, modifying and terminating communication sessions with one or more participants. These sessions include Internet multimedia conferences, Internet telephone calls and multimedia distribution. Members in a session can communicate via multicast or via a mesh of unicast relations, or a combination of these.
A SIP session can include both persons and automated systems such as a media storage service. A SIP session can include both unicast and multicast sessions. A session initiator does not necessarily have to be a member of an initiated session to which SIP is used to initiate. SIP transparently supports name mapping and redirection services, allowing the implementation of ISDN and Intelligent Network telephony subscriber services. These facilities also enable personal mobility.
In the parlance of telecommunications intelligent network services, personal mobility is defined as the ability of end users to originate and receive calls and access subscribed telecommunication services on any terminal in any location, and the ability of the network to identify end users as they move. Personal mobility is based on unique identification numbering and compliments terminal mobility, which enables an end terminal to be moved from one sub-net to another.
SIP is designed as part of the well-known IETF multimedia data and control architecture, which is currently incorporating protocols such as RSVP for reserving network resources; the real-time transport protocol (RTP) for transporting real-time data and providing QoS feedback; the real-time streaming protocol (RTSP) for controlling delivery of streaming media; the session announcement protocol (SAP) for advertising multimedia sessions via multicast; and the session description protocol (SDP) for describing multimedia sessions.
It is known to the inventors that SIP can be used in conjunction with other call setup and signaling protocols. In this mode, an end system uses SIP exchanges to determine the appropriate end system address and protocol from a given address that is protocol-independent. For example, SIP could be used to determine that the party can be reached via H.323, obtain the H.245 gateway and user address and then use H.225.0 to establish a call, for example. In another example, SIP might be used to determine that a call recipient is reachable via the PSTN and indicate the phone number to be called, possibly suggesting an Internet-to-PSTN gateway to be used.
A software suite is known to the inventor for routing communication events over a data-packet-network using an IP session initiation and signaling protocol. The software suite is described with reference to U.S. patent application Ser. No. 09/927,301 listed in the cross-reference section above.
The software suite comprises a server application for computing and serving routing determinations per request, a session management application for initiating and managing routed and established session events, a parsing application for parsing request data received under session initiation protocol and a conversion application for converting data received under session initiation protocol into a routing request. All received communication requests for routing are in the form of the session initiation protocol wherein they are parsed and converted into routing requests processed by the server application and routed to determined destinations. In practice, events are established as session events conducted under the session initiation and management protocol.
The system above is used chiefly within a communication center and uses a proxy server to manage conversion between SIP protocol and T-server routing protocol. It has occurred to the inventor that many organizations that host communication centers host multiple center sites both based in telephony and data networks. It would be desirable to enable seamless communication event transfers including final destination routing between two or more sites that are network connected using un-modified standard protocols including SIP and instant message (IM)-based protocols.
What is therefore clearly needed is a routing system that can transfer communication events seamlessly among multiple communication sites including final agent-level routing of those events using standard SIP and IM-based protocols and preserving complex routing protocols for final agent-level routing within the center site that is the destination for the transfer.